[asterisk-users] RTP traffic not being forwarded

[Ryan Newington]

Hi Vivek,

I'm not sure what you mean, could you explain further?

Regards

Ryan


From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vivek Shrivastava
Sent: Monday, 12 November 2007 1:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded

Hi Ryan,

I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf)

Regards,

Vivek


On 11/11/07, Ryan Newington <ryan at lithiumblue.com<mailto:ryan at lithiumblue.com>> wrote:

Hi Vivek,



The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000.

I can change the port range on the media server, asterisk and the device, but neither seems to help.



My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk)



SIP Phone <---> Media Gateway ---> Asterisk <--- SIP Phone



Ryan





From: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Vivek Shrivastava
Sent: Sunday, 11 November 2007 5:19 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded



Hi Ryan,



Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking

random port selection option on the device/softphone may help.



--Vivek



On 11/10/07, Ryan Newington <ryan at lithiumblue.com<mailto:ryan at lithiumblue.com>> wrote:

Hi Luki,

Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success.

Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway.

SIP Phone <-> Media Gateway <-> Asterisk <-> SIP Phone

An asterisk internal call will work fine. Eg;

SIP Phone <-> Asterisk <-> SIP Phone

Regards

Ryan



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto: asterisk-users-bounces at lists.digium.com <mailto:asterisk-users-bounces at lists.digium.com> ] On Behalf Of Luki
Sent: Sunday, 11 November 2007 12:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTP traffic not being forwarded

> When using 'rtp debug' on the asterisk console, it shows that it is
> receiving traffic from one endpoint, but not the other. A wireshark trace
> reveals it is actually receiving traffic from both ends.

Sounds like a firewall issue. Wireshark shows what's "on the wire",
i.e. before iptables. The packets are being dropped for whatever
reason and never reach the asterisk process. Check your iptables and
RTP port range, and perhaps try changing it.

Luki

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